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QUIC Loss Detection and Congestion ControlGooglejri@google.comGoogleianswett@google.comTransport
QUICThis document describes loss detection and congestion control mechanisms for
QUIC.Discussion of this draft takes place on the QUIC working group mailing list
(quic@ietf.org), which is archived at
https://mailarchive.ietf.org/arch/search/?email_list=quic.Working Group information can be found at https://github.com/quicwg; source
code and issues list for this draft can be found at
https://github.com/quicwg/base-drafts/labels/recovery.QUIC is a new multiplexed and secure transport atop UDP. QUIC builds on decades
of transport and security experience, and implements mechanisms that make it
attractive as a modern general-purpose transport. The QUIC protocol is
described in .QUIC implements the spirit of known TCP loss recovery mechanisms, described in
RFCs, various Internet-drafts, and also those prevalent in the Linux TCP
implementation. This document describes QUIC congestion control and loss
recovery, and where applicable, attributes the TCP equivalent in RFCs,
Internet-drafts, academic papers, and/or TCP implementations.The words “MUST”, “MUST NOT”, “SHOULD”, and “MAY” are used in this document.
It’s not shouting; when they are capitalized, they have the special meaning
defined in .All transmissions in QUIC are sent with a packet-level header, which includes a
packet sequence number (referred to below as a packet number). These packet
numbers never repeat in the lifetime of a connection, and are monotonically
increasing, which makes duplicate detection trivial. This fundamental design
decision obviates the need for disambiguating between transmissions and
retransmissions and eliminates significant complexity from QUIC’s interpretation
of TCP loss detection mechanisms.Every packet may contain several frames. We outline the frames that are
important to the loss detection and congestion control machinery below.Retransmittable frames are frames requiring reliable delivery. The most
common are STREAM frames, which typically contain application data.Crypto handshake data is sent on stream 0, and uses the reliability
machinery of QUIC underneath.ACK frames contain acknowledgment information. QUIC uses a SACK-based
scheme, where acks express up to 256 ranges. The ACK frame also includes a
receive timestamp for each packet newly acked.Readers familiar with TCP’s loss detection and congestion control will find
algorithms here that parallel well-known TCP ones. Protocol differences between
QUIC and TCP however contribute to algorithmic differences. We briefly describe
these protocol differences below.TCP conflates transmission sequence number at the sender with delivery sequence
number at the receiver, which results in retransmissions of the same data
carrying the same sequence number, and consequently to problems caused by
“retransmission ambiguity”. QUIC separates the two: QUIC uses a packet sequence
number (referred to as the “packet number”) for transmissions, and any data that
is to be delivered to the receiving application(s) is sent in one or more
streams, with stream offsets encoded within STREAM frames inside of packets that
determine delivery order.QUIC’s packet number is strictly increasing, and directly encodes transmission
order. A higher QUIC packet number signifies that the packet was sent later,
and a lower QUIC packet number signifies that the packet was sent earlier. When
a packet containing frames is deemed lost, QUIC rebundles necessary frames in a
new packet with a new packet number, removing ambiguity about which packet is
acknowledged when an ACK is received. Consequently, more accurate RTT
measurements can be made, spurious retransmissions are trivially detected, and
mechanisms such as Fast Retransmit can be applied universally, based only on
packet number.This design point significantly simplifies loss detection mechanisms for QUIC.
Most TCP mechanisms implicitly attempt to infer transmission ordering based on
TCP sequence numbers - a non-trivial task, especially when TCP timestamps are
not available.QUIC ACKs contain information that is equivalent to TCP SACK, but QUIC does not
allow any acked packet to be reneged, greatly simplifying implementations on
both sides and reducing memory pressure on the sender.QUIC supports up to 256 ACK ranges, opposed to TCP’s 3 SACK ranges. In high
loss environments, this speeds recovery.QUIC ACKs explicitly encode the delay incurred at the receiver between when a
packet is received and when the corresponding ACK is sent. This allows the
receiver of the ACK to adjust for receiver delays, specifically the delayed ack
timer, when estimating the path RTT. This mechanism also allows a receiver to
measure and report the delay from when a packet was received by the OS kernel,
which is useful in receivers which may incur delays such as context-switch
latency before a userspace QUIC receiver processes a received packet.QUIC uses a combination of ack information and alarms to detect lost packets.
An unacknowledged QUIC packet is marked as lost in one of the following ways:A packet is marked as lost if at least one packet that was sent a threshold
number of packets (kReorderingThreshold) after it has been
acknowledged. This indicates that the unacknowledged packet is either lost
or reordered beyond the specified threshold. This mechanism combines both
TCP’s FastRetransmit and FACK mechanisms.If a packet is near the tail, where fewer than kReorderingThreshold packets
are sent after it, the sender cannot expect to detect loss based on the
previous mechanism. In this case, a sender uses both ack information and an
alarm to detect loss. Specifically, when the last sent packet is
acknowledged, the sender waits a short period of time to allow for
reordering and then marks any unacknowledged packets as lost. This mechanism
is based on the Linux implementation of TCP Early Retransmit.If a packet is sent at the tail, there are no packets sent after it, and the
sender cannot use ack information to detect its loss. The sender therefore
relies on an alarm to detect such tail losses. This mechanism is based on
TCP’s Tail Loss Probe.If all else fails, a Retransmission Timeout (RTO) alarm is always set when
any retransmittable packet is outstanding. When this alarm fires, all
unacknowledged packets are marked as lost.Instead of a packet threshold to tolerate reordering, a QUIC sender may use
a time threshold. This allows for senders to be tolerant of short periods of
significant reordering. In this mechanism, a QUIC sender marks a packet as
lost when a packet larger than it is acknowledged and a threshold amount of
time has passed since the packet was sent.Handshake packets, which contain STREAM frames for stream 0, are
critical to QUIC transport and crypto negotiation, so a separate alarm
period is used for them.Constants used in loss recovery are based on a combination of RFCs, papers,
and common practice. Some may need to be changed or negotiated in order to
better suit a variety of environments.
Maximum number of tail loss probes before an RTO fires.
Maximum reordering in packet number space before FACK style loss detection
considers a packet lost.
Maximum reordering in time space before time based loss detection considers
a packet lost. In fraction of an RTT.
Minimum time in the future a tail loss probe alarm may be set for.
Minimum time in the future an RTO alarm may be set for.
The length of the peer’s delayed ack timer.
The default RTT used before an RTT sample is taken.Variables required to implement the congestion control mechanisms
are described in this section.
Multi-modal alarm used for loss detection.
The number of times the handshake packets have been
retransmitted without receiving an ack.
The number of times a tail loss probe has been sent without
receiving an ack.
The number of times an rto has been sent without receiving an ack.
The last packet number sent prior to the first retransmission
timeout.
The smoothed RTT of the connection, computed as described in
The RTT variance, computed as described in
The largest delta between the largest acked
retransmittable packet and a packet containing retransmittable frames before
it’s declared lost.
The reordering window as a fraction of max(smoothed_rtt, latest_rtt).
The time at which the next packet will be considered lost based on early
transmit or exceeding the reordering window in time.
An association of packet numbers to information about them, including a number
field indicating the packet number, a time field indicating the time a packet
was sent, and a bytes field indicating the packet’s size. sent_packets is
ordered by packet number, and packets remain in sent_packets until
acknowledged or lost.At the beginning of the connection, initialize the loss detection variables as
follows:After any packet is sent, be it a new transmission or a rebundled transmission,
the following OnPacketSent function is called. The parameters to OnPacketSent
are as follows:packet_number: The packet number of the sent packet.is_retransmittable: A boolean that indicates whether the packet contains at
least one frame requiring reliable deliver. The retransmittability of various
QUIC frames is described in . If false, it is still
acceptable for an ack to be received for this packet. However, a caller MUST
NOT set is_retransmittable to true if an ack is not expected.sent_bytes: The number of bytes sent in the packet.Pseudocode for OnPacketSent follows:When an ack is received, it may acknowledge 0 or more packets.The sender MUST abort the connection if it receives an ACK for a packet it
never sent, see .Pseudocode for OnAckReceived and UpdateRtt follow: ack.ack_delay):
rtt_sample -= ack.delay
UpdateRtt(rtt_sample)
// The sender may skip packets for detecting optimistic ACKs
if (packets acked that the sender skipped):
abortConnection()
// Find all newly acked packets.
for acked_packet in DetermineNewlyAckedPackets():
OnPacketAcked(acked_packet.packet_number)
DetectLostPackets(ack.largest_acked_packet)
SetLossDetectionAlarm()
UpdateRtt(rtt_sample):
// Based on {{RFC6298}}.
if (smoothed_rtt == 0):
smoothed_rtt = rtt_sample
rttvar = rtt_sample / 2
else:
rttvar = 3/4 * rttvar + 1/4 * (smoothed_rtt - rtt_sample)
smoothed_rtt = 7/8 * smoothed_rtt + 1/8 * rtt_sample
]]>When a packet is acked for the first time, the following OnPacketAcked function
is called. Note that a single ACK frame may newly acknowledge several packets.
OnPacketAcked must be called once for each of these newly acked packets.OnPacketAcked takes one parameter, acked_packet, which is the packet number of
the newly acked packet, and returns a list of packet numbers that are detected
as lost.If this is the first acknowledgement following RTO, check if the smallest newly
acknowledged packet is one sent by the RTO, and if so, inform congestion control
of a verified RTO, similar to F-RTO Pseudocode for OnPacketAcked follows: 0 &&
acked_packet_number > largest_sent_before_rto)
OnRetransmissionTimeoutVerified()
handshake_count = 0
tlp_count = 0
rto_count = 0
sent_packets.remove(acked_packet_number)
]]>QUIC loss detection uses a single alarm for all timer-based loss detection. The
duration of the alarm is based on the alarm’s mode, which is set in the packet
and timer events further below. The function SetLossDetectionAlarm defined
below shows how the single timer is set based on the alarm mode.The initial flight has no prior RTT sample. A client SHOULD remember
the previous RTT it observed when resumption is attempted and use that for an
initial RTT value. If no previous RTT is available, the initial RTT defaults
to 200ms.Endpoints MUST retransmit handshake frames if not acknowledged within a
time limit. This time limit will start as the largest of twice the rtt value
and MinTLPTimeout. Each consecutive handshake retransmission doubles the
time limit, until an acknowledgement is received.Handshake frames may be cancelled by handshake state transitions. In
particular, all non-protected frames SHOULD be no longer be transmitted once
packet protection is available.When stateless rejects are in use, the connection is considered immediately
closed once a reject is sent, so no timer is set to retransmit the reject.Version negotiation packets are always stateless, and MUST be sent once per
per handshake packet that uses an unsupported QUIC version, and MAY be sent
in response to 0RTT packets.Tail loss probes and
retransmission timeouts are an alarm based mechanism to recover
from cases when there are outstanding retransmittable packets, but an
acknowledgement has not been received in a timely manner.Early retransmit is implemented with a 1/4 RTT timer. It is
part of QUIC’s time based loss detection, but is always enabled, even when
only packet reordering loss detection is enabled.Pseudocode for SetLossDetectionAlarm follows:QUIC uses one loss recovery alarm, which when set, can be in one of several
modes. When the alarm fires, the mode determines the action to be performed.Pseudocode for OnLossDetectionAlarm follows:Packets in QUIC are only considered lost once a larger packet number is
acknowledged. DetectLostPackets is called every time an ack is received.
If the loss detection alarm fires and the loss_time is set, the previous
largest acked packet is supplied.The receiver MUST ignore unprotected packets that ack protected packets.
The receiver MUST trust protected acks for unprotected packets, however. Aside
from this, loss detection for handshake packets when an ack is processed is
identical to other packets.DetectLostPackets takes one parameter, acked, which is the largest acked packet.Pseudocode for DetectLostPackets follows: delay_until_lost):
lost_packets.insert(unacked)
else if (packet_delta > reordering_threshold)
lost_packets.insert(unacked)
else if (loss_time == 0 && delay_until_lost != infinite):
loss_time = now() + delay_until_lost - time_since_sent
// Inform the congestion controller of lost packets and
// lets it decide whether to retransmit immediately.
if (!lost_packets.empty())
OnPacketsLost(lost_packets)
foreach (packet in lost_packets)
sent_packets.remove(packet.packet_number)
]]>The majority of constants were derived from best common practices among widely
deployed TCP implementations on the internet. Exceptions follow.A shorter delayed ack time of 25ms was chosen because longer delayed acks can
delay loss recovery and for the small number of connections where less than
packet per 25ms is delivered, acking every packet is beneficial to congestion
control and loss recovery.The default initial RTT of 100ms was chosen because it is slightly higher than
both the median and mean min_rtt typically observed on the public internet.QUIC’s congestion control is based on TCP NewReno
congestion control to determine the congestion window and pacing rate.QUIC begins every connection in slow start and exits slow start upon
loss. While in slow start, QUIC increases the congestion window by the
number of acknowledged bytes when each ack is processed.Recovery is a period of time beginning with detection of a lost packet.
It ends when all packets outstanding at the time recovery began have been
acknowledged or lost. During recovery, the congestion window is not
increased or decreased.Constants used in congestion control are based on a combination of RFCs,
papers, and common practice. Some may need to be changed or negotiated
in order to better suit a variety of environments.
The default max packet size used for calculating default and minimum
congestion windows.
Default limit on the amount of outstanding data in bytes.
Default minimum congestion window.
Reduction in congestion window when a new loss event is detected.Variables required to implement the congestion control mechanisms
are described in this section.
The sum of the size in bytes of all sent packets that contain at least
one retransmittable frame, and have not been acked or declared lost.
Maximum number of bytes in flight that may be sent.
The packet number after which QUIC will no longer be in recovery.
Slow start threshold in bytes. When the congestion window is below
ssthresh, it grows by the number of bytes acknowledged for each ack.At the beginning of the connection, initialize the loss detection variables as
follows:Invoked at the same time loss detection’s OnPacketAcked is called and
supplied with the acked_packet from sent_packets.Pseudocode for OnPacketAcked follows:Invoked by loss detection from DetectLostPackets when new packets
are detected lost.QUIC decreases the congestion window to the minimum value once the
retransmission timeout has been confirmed to not be spurious when
the first post-RTO acknowledgement is processed.QUIC sends a packet if there is available congestion window and
sending the packet does not exceed the pacing rate.TimeToSend returns infinite if the congestion controller is
congestion window limited, a time in the past if the packet can be
sent immediately, and a time in the future if sending is pacing
limited. congestion_window)
return infinite
return time_of_last_sent_packet +
(packet_size * smoothed_rtt) / congestion_window
]]>This document has no IANA actions. Yet.QUIC: A UDP-Based Multiplexed and Secure TransportGoogleMozillaKey words for use in RFCs to Indicate Requirement LevelsIn many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.Computing TCP's Retransmission TimerThis document defines the standard algorithm that Transmission Control Protocol (TCP) senders are required to use to compute and manage their retransmission timer. It expands on the discussion in Section 4.2.3.1 of RFC 1122 and upgrades the requirement of supporting the algorithm from a SHOULD to a MUST. This document obsoletes RFC 2988. [STANDARDS-TRACK]Forward RTO-Recovery (F-RTO): An Algorithm for Detecting Spurious Retransmission Timeouts with TCPThe purpose of this document is to move the F-RTO (Forward RTO-Recovery) functionality for TCP in RFC 4138 from Experimental to Standards Track status. The F-RTO support for Stream Control Transmission Protocol (SCTP) in RFC 4138 remains with Experimental status. See Appendix B for the differences between this document and RFC 4138.Spurious retransmission timeouts cause suboptimal TCP performance because they often result in unnecessary retransmission of the last window of data. This document describes the F-RTO detection algorithm for detecting spurious TCP retransmission timeouts. F-RTO is a TCP sender-only algorithm that does not require any TCP options to operate. After retransmitting the first unacknowledged segment triggered by a timeout, the F-RTO algorithm of the TCP sender monitors the incoming acknowledgments to determine whether the timeout was spurious. It then decides whether to send new segments or retransmit unacknowledged segments. The algorithm effectively helps to avoid additional unnecessary retransmissions and thereby improves TCP performance in the case of a spurious timeout. [STANDARDS-TRACK]Tail Loss Probe (TLP): An Algorithm for Fast Recovery of Tail LossesRetransmission timeouts are detrimental to application latency, especially for short transfers such as Web transactions where timeouts can often take longer than all of the rest of a transaction. The primary cause of retransmission timeouts are lost segments at the tail of transactions. This document describes an experimental algorithm for TCP to quickly recover lost segments at the end of transactions or when an entire window of data or acknowledgments are lost. Tail Loss Probe (TLP) is a sender-only algorithm that allows the transport to recover tail losses through fast recovery as opposed to lengthy retransmission timeouts. If a connection is not receiving any acknowledgments for a certain period of time, TLP transmits the last unacknowledged segment (loss probe). In the event of a tail loss in the original transmissions, the acknowledgment from the loss probe triggers SACK/FACK based fast recovery. TLP effectively avoids long timeouts and thereby improves TCP performance.Early Retransmit for TCP and Stream Control Transmission Protocol (SCTP)This document proposes a new mechanism for TCP and Stream Control Transmission Protocol (SCTP) that can be used to recover lost segments when a connection's congestion window is small. The "Early Retransmit" mechanism allows the transport to reduce, in certain special circumstances, the number of duplicate acknowledgments required to trigger a fast retransmission. This allows the transport to use fast retransmit to recover segment losses that would otherwise require a lengthy retransmission timeout. [STANDARDS-TRACK]The NewReno Modification to TCP's Fast Recovery AlgorithmRFC 5681 documents the following four intertwined TCP congestion control algorithms: slow start, congestion avoidance, fast retransmit, and fast recovery. RFC 5681 explicitly allows certain modifications of these algorithms, including modifications that use the TCP Selective Acknowledgment (SACK) option (RFC 2883), and modifications that respond to "partial acknowledgments" (ACKs that cover new data, but not all the data outstanding when loss was detected) in the absence of SACK. This document describes a specific algorithm for responding to partial acknowledgments, referred to as "NewReno". This response to partial acknowledgments was first proposed by Janey Hoe. This document obsoletes RFC 3782. [STANDARDS-TRACK]RFC Editor’s Note: Please remove this section prior to
publication of a final version of this document.Integrate F-RTO (#544, #409)Add congestion control (#545, #395)Require connection abort if a skipped packet was acknowledged (#415)Simplify RTO calculations (#142, #417)Overview added to loss detectionChanges initial default RTT to 100msAdded time-based loss detection and fixes early retransmitClarified loss recovery for handshake packetsFixed references and made TCP references informativeImproved description of constants and ACK behaviorAdopted as base for draft-ietf-quic-recoveryUpdated authors/editors listAdded table of contents